default 'sipstorecause' to no

We've decided to disable this feature by default in future 1.8 versions.  This
would be an unexpected behavior change for anyone depending on that SIP_CAUSE
update in their dialplan.

Please refer to the asterisk-dev mailing list more information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html

(issue AST-580)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333009 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index 1c5baf9f54f1b10677311411935b494c27fb4e0c..55752ecfc6489cfe73704c6f2605da0883d9b94a 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -44,7 +44,7 @@ SIP Changes
  * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
    response.  This permits the master channel to know how each channel dialled
    in a multi-channel setup resolved in an individual way. This carries a
-   performance penalty and can be disabled in sip.conf using the
+   performance penalty and must be enabled in sip.conf using the
    'storesipcause' option.
  * Added 'externtcpport' and 'externtlsport' options to allow custom port
    configuration for the externip and externhost options when tcp or tls is used.

diff --git a/UPGRADE.txt b/UPGRADE.txt
index 8d9ce052195b0fe74c06024c0490f54a82c3de78..72dca37bd51d7b226ea128233d05b07fb6751c59 100644 (file)
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -20,6 +20,10 @@
 
 From 1.6.2 to 1.8:
 
+* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
+  This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
+  This carries a performance penalty.
+
 * Asterisk now requires libpri 1.4.11+ for PRI support.
 
 * A couple of CLI commands in res_ais were changed back to their original form:

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 8ffcfbb415d245d95da37ee38d9aad1de3beb1ed..02da6814efa54d17e4daae5201c6c2b6590d98e5 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -221,7 +221,7 @@
 #define DEFAULT_SDPOWNER   "root"          /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
 #define DEFAULT_ENGINE     "asterisk"      /*!< Default RTP engine to use for sessions */
 #define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
-#define DEFAULT_STORE_SIP_CAUSE TRUE      /*!< Store HASH(SIP_CAUSE,<channel name>) for channels by default */
+#define DEFAULT_STORE_SIP_CAUSE FALSE      /*!< Don't store HASH(SIP_CAUSE,<channel name>) for channels by default */
 #endif
 /*@}*/
 

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 2ad86474799b5cdaea6c8bb79fe1ea46b65ca32a..f2b9a2e56bc0311f0c1e8e57eeb9d7b316fc2f58 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1007,14 +1007,13 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
 
 ;----------------------------- SIP_CAUSE reporting ---------------------------------
-; storesipcause = yes         ; This option causes chan_sip to set the
+; storesipcause = no          ; This option causes chan_sip to set the
                              ; HASH(SIP_CAUSE,<channel name>) channel variable
                              ; to the value of the last sip response.
                              ; WARNING: enabling this option carries a
                              ; significant performance burden. It should only
-                             ; be used in low call volume situations. For
-                             ; historical reasons, this option defaults to
-                             ; "yes".
+                             ; be used in low call volume situations. This
+                              ; option defaults to "no".
 
 ;-----------------------------------------------------------------------------------